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swresample.c
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1 /*
2  * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
3  *
4  * This file is part of libswresample
5  *
6  * libswresample is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * libswresample is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with libswresample; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include "libavutil/opt.h"
22 #include "swresample_internal.h"
23 #include "audioconvert.h"
24 #include "libavutil/avassert.h"
26 
27 #include <float.h>
28 
29 #define ALIGN 32
30 
31 #include "libavutil/ffversion.h"
32 const char swr_ffversion[] = "FFmpeg version " FFMPEG_VERSION;
33 
34 unsigned swresample_version(void)
35 {
38 }
39 
40 const char *swresample_configuration(void)
41 {
42  return FFMPEG_CONFIGURATION;
43 }
44 
45 const char *swresample_license(void)
46 {
47 #define LICENSE_PREFIX "libswresample license: "
48  return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1;
49 }
50 
52  if(!s || s->in_convert) // s needs to be allocated but not initialized
53  return AVERROR(EINVAL);
55  return 0;
56 }
57 
61  int log_offset, void *log_ctx){
62  if(!s) s= swr_alloc();
63  if(!s) return NULL;
64 
65  s->log_level_offset= log_offset;
66  s->log_ctx= log_ctx;
67 
68  if (av_opt_set_int(s, "ocl", out_ch_layout, 0) < 0)
69  goto fail;
70 
71  if (av_opt_set_int(s, "osf", out_sample_fmt, 0) < 0)
72  goto fail;
73 
74  if (av_opt_set_int(s, "osr", out_sample_rate, 0) < 0)
75  goto fail;
76 
77  if (av_opt_set_int(s, "icl", in_ch_layout, 0) < 0)
78  goto fail;
79 
80  if (av_opt_set_int(s, "isf", in_sample_fmt, 0) < 0)
81  goto fail;
82 
83  if (av_opt_set_int(s, "isr", in_sample_rate, 0) < 0)
84  goto fail;
85 
86  if (av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0) < 0)
87  goto fail;
88 
89  if (av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0) < 0)
90  goto fail;
91 
93  goto fail;
94 
95  av_opt_set_int(s, "uch", 0, 0);
96  return s;
97 fail:
98  av_log(s, AV_LOG_ERROR, "Failed to set option\n");
99  swr_free(&s);
100  return NULL;
101 }
102 
104  a->fmt = fmt;
105  a->bps = av_get_bytes_per_sample(fmt);
107  if (a->ch_count == 1)
108  a->planar = 1;
109 }
110 
111 static void free_temp(AudioData *a){
112  av_free(a->data);
113  memset(a, 0, sizeof(*a));
114 }
115 
116 static void clear_context(SwrContext *s){
117  s->in_buffer_index= 0;
118  s->in_buffer_count= 0;
120  memset(s->in.ch, 0, sizeof(s->in.ch));
121  memset(s->out.ch, 0, sizeof(s->out.ch));
122  free_temp(&s->postin);
123  free_temp(&s->midbuf);
124  free_temp(&s->preout);
125  free_temp(&s->in_buffer);
126  free_temp(&s->silence);
127  free_temp(&s->drop_temp);
128  free_temp(&s->dither.noise);
129  free_temp(&s->dither.temp);
134 
135  s->flushed = 0;
136 }
137 
139  SwrContext *s= *ss;
140  if(s){
141  clear_context(s);
142  if (s->resampler)
143  s->resampler->free(&s->resample);
144  }
145 
146  av_freep(ss);
147 }
148 
150  clear_context(s);
151 }
152 
154  int ret;
155 
156  clear_context(s);
157 
158  if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){
159  av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt);
160  return AVERROR(EINVAL);
161  }
163  av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt);
164  return AVERROR(EINVAL);
165  }
166 
168  av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout);
169  s->in_ch_layout = 0;
170  }
171 
173  av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout);
174  s->out_ch_layout = 0;
175  }
176 
177  switch(s->engine){
178 #if CONFIG_LIBSOXR
179  case SWR_ENGINE_SOXR: s->resampler = &swri_soxr_resampler; break;
180 #endif
181  case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break;
182  default:
183  av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n");
184  return AVERROR(EINVAL);
185  }
186 
187  if(!s->used_ch_count)
188  s->used_ch_count= s->in.ch_count;
189 
191  av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n");
192  s-> in_ch_layout= 0;
193  }
194 
195  if(!s-> in_ch_layout)
197  if(!s->out_ch_layout)
199 
200  s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 ||
201  s->rematrix_custom;
202 
208  && !s->rematrix
209  && s->engine != SWR_ENGINE_SOXR){
213  }else{
214  av_log(s, AV_LOG_DEBUG, "Using double precision mode\n");
216  }
217  }
218 
223  av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, S16/S32/FLT/DBL is supported\n", av_get_sample_fmt_name(s->int_sample_fmt));
224  return AVERROR(EINVAL);
225  }
226 
229 
231  if (!s->async && s->min_compensation >= FLT_MAX/2)
232  s->async = 1;
233  s->firstpts =
235  } else
237 
238  if (s->async) {
239  if (s->min_compensation >= FLT_MAX/2)
240  s->min_compensation = 0.001;
241  if (s->async > 1.0001) {
242  s->max_soft_compensation = s->async / (double) s->in_sample_rate;
243  }
244  }
245 
248  }else
249  s->resampler->free(&s->resample);
254  && s->resample){
255  av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n");
256  return -1;
257  }
258 
259 #define RSC 1 //FIXME finetune
260  if(!s-> in.ch_count)
262  if(!s->used_ch_count)
263  s->used_ch_count= s->in.ch_count;
264  if(!s->out.ch_count)
266 
267  if(!s-> in.ch_count){
269  av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n");
270  return -1;
271  }
272 
273  if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) {
274  char l1[1024], l2[1024];
275  av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout);
276  av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout);
277  av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s "
278  "but there is not enough information to do it\n", l1, l2);
279  return -1;
280  }
281 
284  s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0;
285 
286  s->in_buffer= s->in;
287  s->silence = s->in;
288  s->drop_temp= s->out;
289 
290  if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){
292  s-> in_sample_fmt, s-> in.ch_count, NULL, 0);
293  return 0;
294  }
295 
297  s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0);
299  s->int_sample_fmt, s->out.ch_count, NULL, 0);
300 
301  if (!s->in_convert || !s->out_convert)
302  return AVERROR(ENOMEM);
303 
304  s->postin= s->in;
305  s->preout= s->out;
306  s->midbuf= s->in;
307 
308  if(s->channel_map){
309  s->postin.ch_count=
311  if(s->resample)
313  }
314  if(!s->resample_first){
315  s->midbuf.ch_count= s->out.ch_count;
316  if(s->resample)
317  s->in_buffer.ch_count = s->out.ch_count;
318  }
319 
323 
324  if(s->resample){
326  }
327 
328  if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0)
329  return ret;
330 
331  if(s->rematrix || s->dither.method)
332  return swri_rematrix_init(s);
333 
334  return 0;
335 }
336 
338  int i, countb;
339  AudioData old;
340 
341  if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count)
342  return AVERROR(EINVAL);
343 
344  if(a->count >= count)
345  return 0;
346 
347  count*=2;
348 
349  countb= FFALIGN(count*a->bps, ALIGN);
350  old= *a;
351 
352  av_assert0(a->bps);
353  av_assert0(a->ch_count);
354 
355  a->data= av_mallocz(countb*a->ch_count);
356  if(!a->data)
357  return AVERROR(ENOMEM);
358  for(i=0; i<a->ch_count; i++){
359  a->ch[i]= a->data + i*(a->planar ? countb : a->bps);
360  if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps);
361  }
362  if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps);
363  av_freep(&old.data);
364  a->count= count;
365 
366  return 1;
367 }
368 
369 static void copy(AudioData *out, AudioData *in,
370  int count){
371  av_assert0(out->planar == in->planar);
372  av_assert0(out->bps == in->bps);
373  av_assert0(out->ch_count == in->ch_count);
374  if(out->planar){
375  int ch;
376  for(ch=0; ch<out->ch_count; ch++)
377  memcpy(out->ch[ch], in->ch[ch], count*out->bps);
378  }else
379  memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps);
380 }
381 
382 static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){
383  int i;
384  if(!in_arg){
385  memset(out->ch, 0, sizeof(out->ch));
386  }else if(out->planar){
387  for(i=0; i<out->ch_count; i++)
388  out->ch[i]= in_arg[i];
389  }else{
390  for(i=0; i<out->ch_count; i++)
391  out->ch[i]= in_arg[0] + i*out->bps;
392  }
393 }
394 
396  int i;
397  if(out->planar){
398  for(i=0; i<out->ch_count; i++)
399  in_arg[i]= out->ch[i];
400  }else{
401  in_arg[0]= out->ch[0];
402  }
403 }
404 
405 /**
406  *
407  * out may be equal in.
408  */
409 static void buf_set(AudioData *out, AudioData *in, int count){
410  int ch;
411  if(in->planar){
412  for(ch=0; ch<out->ch_count; ch++)
413  out->ch[ch]= in->ch[ch] + count*out->bps;
414  }else{
415  for(ch=out->ch_count-1; ch>=0; ch--)
416  out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps;
417  }
418 }
419 
420 /**
421  *
422  * @return number of samples output per channel
423  */
424 static int resample(SwrContext *s, AudioData *out_param, int out_count,
425  const AudioData * in_param, int in_count){
426  AudioData in, out, tmp;
427  int ret_sum=0;
428  int border=0;
429  int padless = ARCH_X86 && s->engine == SWR_ENGINE_SWR ? 7 : 0;
430 
431  av_assert1(s->in_buffer.ch_count == in_param->ch_count);
432  av_assert1(s->in_buffer.planar == in_param->planar);
433  av_assert1(s->in_buffer.fmt == in_param->fmt);
434 
435  tmp=out=*out_param;
436  in = *in_param;
437 
438  border = s->resampler->invert_initial_buffer(s->resample, &s->in_buffer,
439  &in, in_count, &s->in_buffer_index, &s->in_buffer_count);
440  if (border == INT_MAX) {
441  return 0;
442  } else if (border < 0) {
443  return border;
444  } else if (border) {
445  buf_set(&in, &in, border);
446  in_count -= border;
447  s->resample_in_constraint = 0;
448  }
449 
450  do{
451  int ret, size, consumed;
453  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
454  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed);
455  out_count -= ret;
456  ret_sum += ret;
457  buf_set(&out, &out, ret);
458  s->in_buffer_count -= consumed;
459  s->in_buffer_index += consumed;
460 
461  if(!in_count)
462  break;
463  if(s->in_buffer_count <= border){
464  buf_set(&in, &in, -s->in_buffer_count);
465  in_count += s->in_buffer_count;
466  s->in_buffer_count=0;
467  s->in_buffer_index=0;
468  border = 0;
469  }
470  }
471 
472  if((s->flushed || in_count > padless) && !s->in_buffer_count){
473  s->in_buffer_index=0;
474  ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, FFMAX(in_count-padless, 0), &consumed);
475  out_count -= ret;
476  ret_sum += ret;
477  buf_set(&out, &out, ret);
478  in_count -= consumed;
479  buf_set(&in, &in, consumed);
480  }
481 
482  //TODO is this check sane considering the advanced copy avoidance below
483  size= s->in_buffer_index + s->in_buffer_count + in_count;
484  if( size > s->in_buffer.count
485  && s->in_buffer_count + in_count <= s->in_buffer_index){
486  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
487  copy(&s->in_buffer, &tmp, s->in_buffer_count);
488  s->in_buffer_index=0;
489  }else
490  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
491  return ret;
492 
493  if(in_count){
494  int count= in_count;
495  if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2;
496 
497  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
498  copy(&tmp, &in, /*in_*/count);
499  s->in_buffer_count += count;
500  in_count -= count;
501  border += count;
502  buf_set(&in, &in, count);
504  if(s->in_buffer_count != count || in_count)
505  continue;
506  if (padless) {
507  padless = 0;
508  continue;
509  }
510  }
511  break;
512  }while(1);
513 
514  s->resample_in_constraint= !!out_count;
515 
516  return ret_sum;
517 }
518 
519 static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count,
520  AudioData *in , int in_count){
522  int ret/*, in_max*/;
523  AudioData preout_tmp, midbuf_tmp;
524 
525  if(s->full_convert){
526  av_assert0(!s->resample);
527  swri_audio_convert(s->full_convert, out, in, in_count);
528  return out_count;
529  }
530 
531 // in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps;
532 // in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count);
533 
534  if((ret=swri_realloc_audio(&s->postin, in_count))<0)
535  return ret;
536  if(s->resample_first){
538  if((ret=swri_realloc_audio(&s->midbuf, out_count))<0)
539  return ret;
540  }else{
542  if((ret=swri_realloc_audio(&s->midbuf, in_count))<0)
543  return ret;
544  }
545  if((ret=swri_realloc_audio(&s->preout, out_count))<0)
546  return ret;
547 
548  postin= &s->postin;
549 
550  midbuf_tmp= s->midbuf;
551  midbuf= &midbuf_tmp;
552  preout_tmp= s->preout;
553  preout= &preout_tmp;
554 
555  if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map)
556  postin= in;
557 
558  if(s->resample_first ? !s->resample : !s->rematrix)
559  midbuf= postin;
560 
561  if(s->resample_first ? !s->rematrix : !s->resample)
562  preout= midbuf;
563 
564  if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar
566  if(preout==in){
567  out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant
568  av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though
569  copy(out, in, out_count);
570  return out_count;
571  }
572  else if(preout==postin) preout= midbuf= postin= out;
573  else if(preout==midbuf) preout= midbuf= out;
574  else preout= out;
575  }
576 
577  if(in != postin){
578  swri_audio_convert(s->in_convert, postin, in, in_count);
579  }
580 
581  if(s->resample_first){
582  if(postin != midbuf)
583  out_count= resample(s, midbuf, out_count, postin, in_count);
584  if(midbuf != preout)
585  swri_rematrix(s, preout, midbuf, out_count, preout==out);
586  }else{
587  if(postin != midbuf)
588  swri_rematrix(s, midbuf, postin, in_count, midbuf==out);
589  if(midbuf != preout)
590  out_count= resample(s, preout, out_count, midbuf, in_count);
591  }
592 
593  if(preout != out && out_count){
594  AudioData *conv_src = preout;
595  if(s->dither.method){
596  int ch;
597  int dither_count= FFMAX(out_count, 1<<16);
598 
599  if (preout == in) {
600  conv_src = &s->dither.temp;
601  if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0)
602  return ret;
603  }
604 
605  if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0)
606  return ret;
607  if(ret)
608  for(ch=0; ch<s->dither.noise.ch_count; ch++)
609  swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt);
610  av_assert0(s->dither.noise.ch_count == preout->ch_count);
611 
612  if(s->dither.noise_pos + out_count > s->dither.noise.count)
613  s->dither.noise_pos = 0;
614 
615  if (s->dither.method < SWR_DITHER_NS){
616  if (s->mix_2_1_simd) {
617  int len1= out_count&~15;
618  int off = len1 * preout->bps;
619 
620  if(len1)
621  for(ch=0; ch<preout->ch_count; ch++)
622  s->mix_2_1_simd(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_simd_one, 0, 0, len1);
623  if(out_count != len1)
624  for(ch=0; ch<preout->ch_count; ch++)
625  s->mix_2_1_f(conv_src->ch[ch] + off, preout->ch[ch] + off, s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos + off + len1, s->native_one, 0, 0, out_count - len1);
626  } else {
627  for(ch=0; ch<preout->ch_count; ch++)
628  s->mix_2_1_f(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_one, 0, 0, out_count);
629  }
630  } else {
631  switch(s->int_sample_fmt) {
632  case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break;
633  case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break;
634  case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break;
635  case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break;
636  }
637  }
638  s->dither.noise_pos += out_count;
639  }
640 //FIXME packed doesn't need more than 1 chan here!
641  swri_audio_convert(s->out_convert, out, conv_src, out_count);
642  }
643  return out_count;
644 }
645 
647  return !!s->in_buffer.ch_count;
648 }
649 
650 int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count,
651  const uint8_t *in_arg [SWR_CH_MAX], int in_count){
652  AudioData * in= &s->in;
653  AudioData *out= &s->out;
654 
655  if (!swr_is_initialized(s)) {
656  av_log(s, AV_LOG_ERROR, "Context has not been initialized\n");
657  return AVERROR(EINVAL);
658  }
659 
660  while(s->drop_output > 0){
661  int ret;
662  uint8_t *tmp_arg[SWR_CH_MAX];
663 #define MAX_DROP_STEP 16384
665  return ret;
666 
667  reversefill_audiodata(&s->drop_temp, tmp_arg);
668  s->drop_output *= -1; //FIXME find a less hackish solution
669  ret = swr_convert(s, tmp_arg, FFMIN(-s->drop_output, MAX_DROP_STEP), in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesn't matter
670  s->drop_output *= -1;
671  in_count = 0;
672  if(ret>0) {
673  s->drop_output -= ret;
674  if (!s->drop_output && !out_arg)
675  return 0;
676  continue;
677  }
678 
680  return 0;
681  }
682 
683  if(!in_arg){
684  if(s->resample){
685  if (!s->flushed)
686  s->resampler->flush(s);
687  s->resample_in_constraint = 0;
688  s->flushed = 1;
689  }else if(!s->in_buffer_count){
690  return 0;
691  }
692  }else
693  fill_audiodata(in , (void*)in_arg);
694 
695  fill_audiodata(out, out_arg);
696 
697  if(s->resample){
698  int ret = swr_convert_internal(s, out, out_count, in, in_count);
699  if(ret>0 && !s->drop_output)
700  s->outpts += ret * (int64_t)s->in_sample_rate;
701  return ret;
702  }else{
703  AudioData tmp= *in;
704  int ret2=0;
705  int ret, size;
706  size = FFMIN(out_count, s->in_buffer_count);
707  if(size){
708  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
709  ret= swr_convert_internal(s, out, size, &tmp, size);
710  if(ret<0)
711  return ret;
712  ret2= ret;
713  s->in_buffer_count -= ret;
714  s->in_buffer_index += ret;
715  buf_set(out, out, ret);
716  out_count -= ret;
717  if(!s->in_buffer_count)
718  s->in_buffer_index = 0;
719  }
720 
721  if(in_count){
722  size= s->in_buffer_index + s->in_buffer_count + in_count - out_count;
723 
724  if(in_count > out_count) { //FIXME move after swr_convert_internal
725  if( size > s->in_buffer.count
726  && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){
727  buf_set(&tmp, &s->in_buffer, s->in_buffer_index);
728  copy(&s->in_buffer, &tmp, s->in_buffer_count);
729  s->in_buffer_index=0;
730  }else
731  if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0)
732  return ret;
733  }
734 
735  if(out_count){
736  size = FFMIN(in_count, out_count);
737  ret= swr_convert_internal(s, out, size, in, size);
738  if(ret<0)
739  return ret;
740  buf_set(in, in, ret);
741  in_count -= ret;
742  ret2 += ret;
743  }
744  if(in_count){
745  buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count);
746  copy(&tmp, in, in_count);
747  s->in_buffer_count += in_count;
748  }
749  }
750  if(ret2>0 && !s->drop_output)
751  s->outpts += ret2 * (int64_t)s->in_sample_rate;
752  return ret2;
753  }
754 }
755 
756 int swr_drop_output(struct SwrContext *s, int count){
757  const uint8_t *tmp_arg[SWR_CH_MAX];
758  s->drop_output += count;
759 
760  if(s->drop_output <= 0)
761  return 0;
762 
763  av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count);
764  return swr_convert(s, NULL, s->drop_output, tmp_arg, 0);
765 }
766 
768  int ret, i;
769  uint8_t *tmp_arg[SWR_CH_MAX];
770 
771  if(count <= 0)
772  return 0;
773 
774 #define MAX_SILENCE_STEP 16384
775  while (count > MAX_SILENCE_STEP) {
776  if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0)
777  return ret;
778  count -= MAX_SILENCE_STEP;
779  }
780 
781  if((ret=swri_realloc_audio(&s->silence, count))<0)
782  return ret;
783 
784  if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) {
785  memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps);
786  } else
787  memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count);
788 
789  reversefill_audiodata(&s->silence, tmp_arg);
790  av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count);
791  ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count);
792  return ret;
793 }
794 
795 int64_t swr_get_delay(struct SwrContext *s, int64_t base){
796  if (s->resampler && s->resample){
797  return s->resampler->get_delay(s, base);
798  }else{
799  return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
800  }
801 }
802 
803 int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
804  int ret;
805 
806  if (!s || compensation_distance < 0)
807  return AVERROR(EINVAL);
808  if (!compensation_distance && sample_delta)
809  return AVERROR(EINVAL);
810  if (!s->resample) {
811  s->flags |= SWR_FLAG_RESAMPLE;
812  ret = swr_init(s);
813  if (ret < 0)
814  return ret;
815  }
816  if (!s->resampler->set_compensation){
817  return AVERROR(EINVAL);
818  }else{
819  return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance);
820  }
821 }
822 
823 int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
824  if(pts == INT64_MIN)
825  return s->outpts;
826 
827  if (s->firstpts == AV_NOPTS_VALUE)
828  s->outpts = s->firstpts = pts;
829 
830  if(s->min_compensation >= FLT_MAX) {
831  return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate));
832  } else {
833  int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts + s->drop_output*(int64_t)s->in_sample_rate;
834  double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
835 
836  if(fabs(fdelta) > s->min_compensation) {
837  if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){
838  int ret;
839  if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate);
840  else ret = swr_drop_output (s, -delta / s-> in_sample_rate);
841  if(ret<0){
842  av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta);
843  }
847  int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ;
848  av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration);
849  swr_set_compensation(s, comp, duration);
850  }
851  }
852 
853  return s->outpts;
854  }
855 }